• Proceedings of the Korean Society of Marine Engineers Conference
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• 한국마린엔지니어링학회 2005년도 후기학술대회논문집
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• pp.65-66
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• 2005

This paper presents an adaptive satellite communication systems to adapt channel environment. High performance coding and modulation techniques are applied to poor channel condition, otherwise, bandwidth efficient coding and modulation techniques are applied to good channel condition to obtain high transmission rate.

• IEIE Transactions on Smart Processing and Computing
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• 제6권4호
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• pp.276-280
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• 2017

This paper presents computationally efficient image compression for wireless transmission of high-definition video, to adaptively utilize available channel bandwidth and improve image quality. The method indirectly predicts an unknown available channel bandwidth by monitoring encoder buffer status, and adaptively controls a quantization parameter to fully utilize the bandwidth. Experimental results show that the proposed method is robust to variations in channel bandwidth.

• IEIE Transactions on Smart Processing and Computing
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• 제5권5호
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• pp.303-309
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• 2016

This paper presents a channel-adaptive rate control algorithm for low delay video coding. The main goal of the proposed method is to adaptively use the unknown available channel bandwidth while reducing the end-to-end delay between encoder and decoder. The key idea of the proposed algorithm is for the status of the encoder buffer to indirectly reflect the mismatch between the available channel bandwidth and the generated bitrate. Hence, the proposed method fully utilizes the unknown available channel bandwidth by monitoring the encoder buffer status. Simulation results show that although the target bitrate mismatches the available channel bandwidth, the encoder efficiently adapts the given available bandwidth to improve the peak signal-to-noise ratio.

• Proceedings of the IEEK Conference
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• 대한전자공학회 2001년도 하계종합학술대회 논문집(1)
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• pp.73-76
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• 2001

A key requirement for packet based wireless communication systems is to provide a high data rate packet service and improved throughput. To achieve a high throughput, adaptive methods for adjustment of the modulation and coding can be used. In this paper, we propose and analyze a scheme which is a combination of an adaptive modulation and coding(AMC) and transmit diversity(TD). Two different TD schemes are analysed: STTD and STD. Proposed system provides significant improvement in the average throughput.

• Journal of electromagnetic engineering and science
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• 제16권2호
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• pp.80-86
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• 2016

This paper proposes the algorithm for forward link adaptive coding and modulation (ACM) and the detailed design for a satellite communication system to improve network reliability and system throughput. In the ACM scheme, the coding and modulation schemes are changed by as much as the channel can provide depending on the quality of the communication link. To implement the forward link ACM system in the Ka-band, channel prediction and modulation/coding decision methods are proposed and simulated. The parameters of the adaptive filter predictor based on the least mean square are optimized, the minimum mean square error of the channel predictor is 0.0608 when step size and the number of filter tap are 0.0001 and 4, respectively. A test-bed is set up to verify the forward link ACM system, and a test is performed using a Ka-band satellite (i.e., Communication, Ocean, and Meteorological Satellite [COMS]). This test verifies that the ACM scheme can increase the system throughput.

• The Journal of the Korea Contents Association
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• 제8권4호
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• pp.55-62
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• 2008

Adaptive modulation and Coding(AMC) scheme is promising technique to support the demands for high data rates and wideband proposed for 4G mobile communication system standards. In this paper, adaptive modulation and coding(AMC) based on OFDM system is analyzed through simulation for single user case and compared with SISO-OFDM and SFBC(Space frequency block coding)-OFDM. The performance analysis in terms of capacity for downlink system environments with different values of constellation size under multipath fading channel is done. The adaptive modulation and coding technique is based on perfect estimation channel. It has been observed that SFBC(Space-frequency block coding)-OFDM system gives better performance in terms of capacity.

• Journal of Communications and Networks
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• 제7권3호
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• pp.243-247
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• 2005

We propose an efficient block least-mean-square (BLMS) adaptive algorithm, in conjunction with error control coding, for direct-sequence code division multiple access (DS-CDMA) systems. The proposed adaptive receiver incorporates decision feedback detection and channel encoding in order to improve the performance of the standard LMS algorithm in convolutionally coded systems. The BLMS algorithm involves two modes of operation: (i) The training mode where an uncoded training sequence is used for initial filter tap-weights adaptation, and (ii) the decision-directed where the filter weights are adapted, using the BLMS algorithm, after decoding/encoding operation. It is shown that the proposed adaptive receiver structure is able to compensate for the signal-to­noise ratio (SNR) loss incurred due to the switching from uncoded training mode to coded decision-directed mode. Our results show that by using the proposed adaptive receiver (with decision feed­back block adaptation) one can achieve a much better performance than both the coded LMS with no decision feedback employed. The convergence behavior of the proposed BLMS receiver is simulated and compared to the standard LMS with and without channel coding. We also examine the steady-state bit-error rate (BER) performance of the proposed adaptive BLMS and standard LMS, both with convolutional coding, where we show that the former is more superior than the latter especially at large SNRs ($SNR\;\geq\;9\;dB$).

• Journal of KIISE:Computing Practices and Letters
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• 제16권10호
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• pp.990-994
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• 2010

Mobile WiMAX does not guarantee the media qualities because it does not consider the characteristics of video coding techniques. In this paper, PC-MCA(Priority-based Combining adaptive Modulation and Coding with ARQ), a priority based channel-adaptive streaming scheme, is proposed to guarantee media qualities. PC-MCA uses QoS scheduler by scheduling priority of the media and differentially controls modulation and coding schemes according to wireless channel condition and frame priorities. It also guarantees multimedia service quality through video decoding reliability.

• Proceedings of the IEEK Conference
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• 대한전자공학회 2000년도 하계종합학술대회 논문집(4)
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• pp.92-97
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• 2000

An AMR(Adaptive Multi-Rate) speech coding algorithm has been adopted as a standard speech codec for IMT-2000. It is based on the algebraic CELP, and consists of eight speech coding modes having the bit rate from 4.75 kbit/s to 12.2 kbit/s. It also contains the VAD(Voice Activity Detector), SCR (Source Controlled Rate) operation, and error concealment scheme for robustness in a radio channel. The bit rate of AMR is changed on a frame basis depending on the channel condition. In this paper, we introduced AMR speech coding algorithm and performed the real-time implementation using TMS320C6201, i.e., a Texas Instrument's fixed-point DSP. With the ANSI C source code released from ETSI and 3GPP, we convert and optimize the program to make it run in real time using the C compiler and assembly language. It is verified that the decoded result of the implemented speech codec on the DSP is identical with the PC simulation result using ANSI C code for test sequences. Also, actual sound input/output test using microphone and speaker demonstrates its proper real-time operation without distortions or delays.

• The Journal of Korean Institute of Communications and Information Sciences
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• 제29권5A호
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• pp.492-501
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• 2004

The performance of mobile communication systems depends on the state of the time-varying multi-path fading channel. To effectively prevent the corruption of video stream and its propagation in spatial and temporal domain, proactive error controls are widely being deployed. Among possible candidates, the rate compatible punctured convolutional (RCPC) code has been widely used for multimedia data, since its rate can be determined flexibly. In this paper, the adaptive channel estimation and the adaptive error correction techniques over the time-varying mobile channel have been proposed. Extensive computer simulations show that the proposed techniques yield the superior performance than the fixed rate system.